-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2963/
-----------------------------------------------------------

(Updated Nov. 27, 2013, 6:36 p.m.)


Status
------

This change has been marked as submitted.


Review request for Asterisk Developers.


Changes
-------

Committed in revision 403207


Repository: Asterisk


Description
-------

chan_pjsip currently supports only one method for handling redirects: It takes 
the user portion of the target and places it into the call forwarding target as 
a local extension. This is fine for calling end-user devices but is not 
suitable for some situations involving other SIP servers (*cough* Microsoft 
Lync *cough*). The attached patch makes the behavior configurable and adds two 
other options: "uri_dialplan" and "uri_pjsip".

The uri_dialplan option returns the URI as the call forwarding target and 
instructs the dial process to dial it using the original endpoint. This is the 
equivalent of the "promiscredir" option in chan_sip.

The uri_pjsip option handles the redirect completely within chan_pjsip itself. 
This allows multiple targets to be tried if need be, and also reduces the 
amount of work the core has to do (no channel teardown and dialing again, the 
same channel is used).

As all of these may be useful for people and implementing them is relatively 
easy I've done so.


Diffs
-----

  /branches/12/res/res_pjsip_session.c 402863 
  /branches/12/res/res_pjsip/pjsip_configuration.c 402863 
  /branches/12/res/res_pjsip.c 402863 
  /branches/12/include/asterisk/res_pjsip.h 402863 

Diff: https://reviewboard.asterisk.org/r/2963/diff/


Testing
-------

Placed calls to a target with each option, confirmed that they work as expected.


Thanks,

Joshua Colp

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to